SHORT QUESTIONS ANSWER________________________

______________________UNIT I____________________
2.Distinguish between energy and power signal.
3.How can we prevent aliasing?
Aliasing is a effect of violation of the Nyquist-Shannon-Sampling-Theory. During the
sampling the base band spectrum of the sampled signal ist mirrored to every multifold of
the sampling frequency. These mirrored spectra are called alias. If the signal spectrum
reaches farther than half the sampling frequency base band spectrum and aliases touch
each other and the base band spectrum gets superimposed by the first alias spectrum. The
easiest way to prevent aliasing is the application of a steep sloped low-pass filter with half
the sampling frequency before the conversion.
4.Classify the signals?
Multichannel Signal:
Signals generated by multiple sources or multiple sensors. Ex: EEG signal.
One dimensional Signals:
A function of single independent variable. Ex: v(t) = V
m
sin(t)
Multi Dimensional Signals:
A function of two or more independent variables.
Ex: Photo 2 dimensional signal
Ex: B/W TV Picture intensity 3 dimensional signal, I(x,y,t)
5.What is a multi channel signal?
Signals generated by multiple sources or multiple sensors. Ex: EEG signal.
6.State analog signal.
The analog signal is a continuous function of an indepentent variable such as time,
space, etc.
14.Find whether the given system is static or dynamic. y(n) = n x(n)+5x
3
(n)
The output y(n) depends on the input at that instant only. Therefore the system
is static.
15.Determine z transform and ROC of the signal {1,2,3,4}
Solution:
The Z-transform of a sequence ,
( ) ( )
3
0
n
x z x n z

( )
0 1 2 3
(0) (1) (2) (3) x z x z x z x z x z

+ + +
Where x(0) = 1
x(1) = 2
x(2) = 3
x(3) = 4
Substitute x(0),x(1), x(2) and x(3) values in above equation,
( )
0 1 2 3
(1) * (2) * (3) * (4) * x z z z z z

+ + +
Answer :
( )
0 1 2 3
2 3 4 x z z z z z

+ + +
16.List the mathematical operations performed on discrete time signals.
Shifting
Time reversal
Shifting after time reversal
Time scaling
aaScalar multiplication
Signal multiplier
Signal addition
17.Find whether the given system is linear or not. Y(n)=n x(n)
Let H be the system represented by the equation , y(u) = n x(n) and the system H
operates on x(n) to produce y(n) = H[x(n)] = n x(n)
Consider two signals x
1
(n) and x
2
(n).
Let y
1
(n) and y
2
(n) be the response of the system H for inputs x
1
(n) and x
2
(n) respectively.
y
1
(n) = H[x
1
(n)] = n x
1
(n) .(1)
y
2
(n) = H[x
1
(n)] = n x
2
(n) .(2)
consider av linear combination of inputs a
1
x
1
(n)+ a
2
x
2
(n). let the response of the system for this
linear combination of inputs be y
3
(n).
y
3
(n) = H[a
1
x
1
(n)+ a
2
x
2
(n)] = n(a
1
x
1
(n)+ a
2
x
2
(n)) = a
1
nx
1
(n)+ a
2
nx
2
(n) (3)
The condition to be satisfied for linearity is, y
3
(n) = a
1
y
1
(n)+ a
2
y
2
(n) .(4)
From equation (1) we get, a
1
y
1
(n) = a
1
nx
1
(n)
From equation (2) we get, a
2
y
2
(n) = a
2
nx
2
(n)
Therefore, a
1
y
1
(n)+ a
2
y
2
(n) = a
1
nx
1
(n)+ a
2
nx
2
(n) (5)
From equations (4) and (5) we can say that the condition for linearity is satisfied.
i.e., y
3
(n) = a
1
y
1
(n)+ a
2
y
2
(n). Hence the system is linear.
18.What is meant by ROC?
The values of z for which the z-transform converges is called region of
convergence (ROC).
19.Define z transform.
The z-transform of the discrete-time signal x[n] is defined to be
where is a complex variable. In polar form z can be expressed as z = re
j
, where r
is the radius of the circle.
20.List the various methods of classifying discrete time system.
Static & Dynamic systems
Time-variant & Time-invariant systems
Causal & Non-causal systems
Linear & Non-linear systems
Stable & Unstable systems
FIR & IIR systems
21.Determine z transform and ROC of the signal {5,6,7,8}
x (n) = {5,6,7,8}
Solution :
The Z-transform of a sequence ,
( ) ( )
0
3
n
x z x n z

( )
3 2 1 0
( 3) ( 2) ( 1) (0) x z x z x z x z x z + + +
Where x(0) = 8
x(-1) = 6
x(-2) = 7
x(-3) = 8
Substitute x(0), x(-1), x(-2) and x(-3) values in above equation,
( )
3 2 1 0
(8) * (7) * (6) * (5) * x z z z z z + + +
Answer :
( )
3 2 1 0
8 7 6 5 x z z z z z + + +
22.What are the various methods of representing discrete time signal?
Graphical representation
Functional representation
Tabular representation
Sequence representation
23.How will you classify the discrete time signal?
Signals can be classified into various types by
i. Nature of the independent variables
ii.Value of the function defining the signals
24.List out some important properties of ROC.
The importance of the ROC cannot be overemphasized. It is part of the Z-transform.
In specifying the Z-transform x( z) of a signal x( n), the ROC must be given -
otherwise, the Z-transform cannot be inverted - in order to re-obtain x( n) from X(z),
the ROC must be given.
Example - Consider two sequences
Here it is important to understand that X(z) Y(z). The ROCs of these two
transforms do not even intersect. Not equal! X( z) + ROC unique x( n).
ROC Shape is either Ring or disk in the z-plane centered at the origin.
Fourier Transform converges absolutely if and only if the ROC of the Z-transform of
x(n) includes unite circle.
ROC cannot contain poles.
ROC Region must be a connected region.
25.Determine the convolution sum of two sequences x(n)={3,2,1,2} and h(n)={1,2,1,2}.
Solution :
H(n) = x(n) * h(n)
h(n)
x(n)
1 2 1 2
3 3 6 3 6
2 2 4 2 4
1 1 2 1 2
2 2 4 2 4
y(0) = 3
y(1) = 2+6 = 8
y(2) = 1+4+3 = 8
y(3) = 2+2+2+6 = 12
y(4) = 4+1+4 = 9
y(5) = 2+2 =4
y(6) = 4
y(n) = {3,8,8,12,9,4,4}
_______________________UNIT II______________________
1. List down any four properties of DTFT.
Linearity
Periodicity
Time shifting
Time reversal
Convolution
Frequency shift
1. Define DFT.
The discrete fourier transform compute the values of the z-transform for
evenly spaced points around the unit circle for a given sequence.
2. State shifting property of DFT.
Let x
p
(n) is a periodic sequence with period N, which is obtained by x(n)
( ) ( )
p
t
x n x n lN

( ) ( )
p
t
x n k x n k lN

3. What is zero padding?

) ( ) ( ]} [ ] [ { 2 2 1 1 2 2 1 1 X a X a n x a n x a F + +
integer an is k where ), ( ) 2 ( X k X +
) ( ]} [ { 0

X n x e F
n j o
) ( ]} [ { X n x F
) ( ) ( ]} [ ] [ { H X n h n x F
) ( ]} [ {

X e k n x F
k j t
t
4. Let the sequence x(n) has a length L. If we want to find the N-point DFT (N-L) of
the sequence x(n). we have to add (N-L) zeros to the sequences x(n). this is known
as zero padding.
Uses :
1. We can get better display of the frequency spectrum.
2. With zero padding, the DFT can be used in linear filtering.
1. Find Fourier transform of a sequence

'

otherwise
n for
0
2 2 1
Solution :
( )
1 2 2
0
for n
x n
otherwise

'

( ) ( )
j n
n
X x n e

( )
2
2
j n
n
X e

( ) ( ) ( ) ( ) ( ) ( )
2 0 2
2 1 0 1 2
j j j j
X x e x e x e x e x e

+ + + +
( )
2 0 2 j j j j
X e xe e e e

+ + + +
2. What is DIT radix-2 FFT?
The DIT radix-2 FFT is an efficient algorithm for computing DFT. The time
domain N-point sequence is decimated into 2-point sequences. The result of 2-
point DFTs are used to compute 4-points DFTs. Two numbers of 2-point DFTs
are combined to get a 4-point DFT. The results of 4-point DFTs are used to
compute 8-point DFTs. Two numbers of 4-point DFTs are combined to get an 8-
point DFT.
7.Find the computation efficiency of 1024 point FFT over 1024 point DFT.
The direct computation of 1024 point DFT requires N
2
= (1024)
2
= 1048576
multiplications.
The 1024 points requires
2
log
2
N
N
2
1024
log 1024 5120
2
multiplications
8.Why FFT is needed?
FFT is algorithms are required for the coefficient computation of DFT.
The direct computation of DFT requires N
2
complex multiplication and N
2
N complex
additions where as FFT requires only
2
log
2
N
N
complex multiplication and
2
log
2
N
N

complex additions. If N increases then the processing speed increases in FFT algorithms.
9.What is the sufficient condition for the existence of DTFT?
The DTFT equation will converge, if x(n) is absolutely summable.
( )
n
x n

<

This is the sufficient and stronger dirichlet condition , the first two dirichlet conditions
of continuous time signals are not applicable in discrete time signal. Some signals are not
absolutly summable, but they are square summable.
10.Distinguish between DTFT and DFT.
Te spectrum of DTFT x() is continuous and if is not convenient to
calculate x() in digital signal processor.
11.How many multiplications and additions are required to compute N-point
DFT using radix-2 FFT.
Number of multiplications
2
log
2
N
N
Number of additions
2
log N N
12.What are the applications of FFT algorithms?
Linear filtering
Correlation
Spectrum analysis
13.Give relationship between DTFT, DFT and Z- transform.
The DTFT X() and z-transform X(z) are related by X() = H(z)|
z=e
j

The DTFT X(k) and DTFT X() are related by X(k) = H()|
=e
j2k/N
14.Write the application of Fourier transforms.
The frequency response of LTI system is given by the fourier transform of the
impulse response of the system.
The ratio of the fourier transform of output to fourier transform of input is the transfer
function of the system is frequency domain.
The response of an LTI system can be easily computed using convolution property of
fourier transform.
15.Determine the DTFT of the sequence x(n) = {1,-1,1,-1).
( ) ( )
j n
n
X x n e

( )
3
0
( )
j n
n
X x n e

( ) ( ) ( ) ( ) ( )
0 2 3
0 1 2 3
j j j
X x e x e x e x e

+ + +
( )
2 3
1
j j j
X e e e

+
16.Draw the radix-2 butterfly diagram for DIT and DIF algorithms.
Basic butterfly of DIT radix -2
Basic butterfly of DIF radix -2
17.Arrange the 8 point sequence x(n)={1,2,3,4,-1,-2,-3,-4} in bit reversed order.
In General order Bit Reverse Bit Reverse order
x(0) 0 0 0 0 0 0 x(0)
x(1) 0 0 1 1 0 0 x(4)
x(2) 0 1 0 0 1 0 x(2)
x(3) 0 1 1 1 1 0 x(6)
x(4) 1 0 0 0 0 1 x(1)
x(5) 1 0 1 1 0 1 x(5)
x(6) 1 1 0 0 1 1 x(3)
x(7) 1 1 1 1 1 1 x(7)
x(n) = { 1, -1, 3, -3, 2, -2, 4, -4 }
18.What are the differences between DIT and DIF algorithms?
The DIT input is bit reversal while the output is in natural order, where as for DIF
input is in natural order while the output is bit reversal.
The DIF butterfly is slightly different from the DIT butterfly. The difference being
that the complex multiplications takes place after the addition , subtraction operation
in DIF.
______________________UNIT-III_______________________
1. Draw the general realization structure in direct form-I of IIR system.
+
z
-1
z
-1
z
-1
z
-1
z
-1
z
-1
+++++++
2. State the condition for the stability of digital filter.
The analog system function H(s) is stable if all its poles lies in the left
half of the s-plane similarly the digital system function H(z) is stable . if it stratifies
the following properties.
1. The j axis in the s-plane map into the unit circle in the z-plane
2. The lest half plane of the s-plane should map into the inside of the unit circle in the
z-plane thus a stable analog filter will be converted into a stable digital filter.
X(z) Y(z) Z
-1
X(z) b
0
b
1
b
M-1
b
M
-a
1
-a
N-1
-a
N
Z
-
2
X(z)
Z
-M
X(z) b
2
Z
-(M-1)
X(z) Z
-1
Y(z) Z
-2
Y(z) -a
2
Z
-(N-1)
Y(z) Z
-N
Y(z)
++++++++
1. Define IIR.
If the system is designed by choosing all the infinite sample of impulse
response then it is called IIR system.
2. Mention any two procedures for digitizing transfer function of an analog filter.
The two important procedures for digitizing the transfer functions of
analog filter are
1.impulse invariant method
2.bilinear transformation method
3. Compare the digital and analog filter.
Sl.no Digital filter Analog filter
1 Operates on digital samples of the
signal
Operates on analog signals
2 It is defined by the linear
difference equation
It is defined by linear
difference
3 It consists of adder, multiplier and
delays implemented in digital logic
It consists of electrical
components like resistors,
capacitors, inductors
4 The filter co efficient are designed
to statisfy the desired frequency
response
The approximation
problem is solved to
statisfy the desired
frequency response
4. Mention the important features of IIR filters.
1. The physically realizable IIR filter does not have linear phase.
2. The IIR filter specification includes the desired characteristics for the magnitude
response only.
7.How bilinear transformation is performed?
The bilinear transformation is performed by letting
1
1
2 1
1
s
T

+
Z
Z
In the analog filter transfer function
i.e H(Z) =H
a
(s)
1
1
2 1
1
s
T

+
Z
Z
5. What are the advantages of bilinear transformation?
1.The bilinear transformation provides one to one mapping.
2.Stable continuous systems can be mapped into realizable, stable digital systems
3.There is no aliasing.
6. List out the basic properties of IIR filters.
1.The imaginary axis in the s-plane should map into the unit circle in the z-plane . this
is will be direct relationship between two frequency variables in the two domain
2.The left half s-plane should map into the inter unit circle in the z-plane . the stable
analog filter will be converted to a stable digital filter.
7. Classify the filters based on frequency response.
Based on the frequency response, the filters can be classified into the four types.
1.Low pass
2.High pass
3.Band pass
4.Band stop filters
10.How will you determine the order N of chebyshev filter?
Calculate a parameter N, using the following equation and correct it to nearest integer

1
]
1

'

1
]
1

1
2 1
2
1
2
2
1
1
cosh
1
1 1
cosh
A
N

8. What are the steps involved in the design of digital IIR filter.
1.The specification of a digital filters are converted into the specification of analog
filter.
2.The analog filters is designed for the specification
3.The analog filter is converted to digital filters by transformation techniques.
9. Give the transformation used in the approximation of derivates.
The analog filter H(S) is transformed to digital filter H(Z) by
H(Z) = H(S)
1
1
s
T

Z
10. Write the properties of Butterworth filter.
1.The magnitude response of a Butterworth filter has a monotonic techniques flat pass
band and stop band
2.The poles of the Butterworth filter lies on a circle .
11. Distinguish between recursive and non recursive realization.
For recursive realization the present output y(n) is a function of past
output and present input . this form corresponds to an infinite impulse response IIR
digital filter
For non recursive realization the current output y(n) is a function of
only past and present input . this form corresponds to an finite impulse response FIR
digital filter.
12. What do you understand by backward difference?
One of the simplest method for converting an analog filter into a digital filter
is to approximate the differential equation by an equivalent differential equation.
i.e d/dt .y(t) t =nT = y(nT) y(nT-T)
T
= y(n) y(n-1)
T
The above equation is known as backward difference
13. Give the magnitude function of Butterworth filter.
The magnitude function of Butterworth filter

1/ 2
2
1
( )
1
N
C
H j
_
_
+

,
,
N = 1,2,3
14. Sketch the mapping of S-plan to Z-plan in bilinear transformation.

Jv

J1
U
-1 +1
-J1
____________________________UNIT IV___________________________
1. List the well known design technique for linear phase FIR filter design.
There are three well known method of design techniques for linear phase FIR filter.
They are:
1. Fourier series method and window method
2. Frequency sampling methods
3. Optional filter design methods.
2. What are the types of digital filter according to their impulse response?
Based on impulse response filter are of two types
IIR filters:
The IIR filter are of recursive type, where by the present output depends on
the present input , past input samples and output samples.
FIR filters:
The FIR filters are of non recursive type, where by the present output sample depends
on the present input , sample and previous input sample.
3. How phase distortion and delay distortion are introduced?
The phase distortion is introduced when the phase characteristics of a filter is not
linear within the desired frequency band .
The delay distortion is introduced when the delay is not a constant within the desired
frequency range.
4. Write the steps involved in FIR filter design.
1.Choose the desired (idea) frequency response H
d
()
2.Take inverse fourier transform of H
d
() to get h
d
(n)
3.Convert the infinite duration h
d
(n)to finite duration sequence h(n).
4.Take Z transform of h(n) to get the transfer function H(Z) of the FIR filters.
5. Compare hamming window with Kaiser window.
Sl.no Hamming window Kaiser window
1 The width of main lobe in
window spectrum is 8/N
The width of main lobe in
window spectrum depends on the
values of and N
2 The max side lobe
magnitude in window
spectrum is fixed at -41dB
The maximum side lobe
magnitude with respect to peak
of main lobe is variable using the
parameter
3 In window spectrum the
side lobe magnitude
remain constant with
increasing
In window spectrum the side
lone magnitude decrease with
increasing
6. Draw the impulse response of an ideal low pass filter.
h
d
(n)
7. What are the advantages of FIR filter?
1. Linear phase FIR filters can be easily designed
2. Efficient realizations of FIR filter exist as both recursive and non recursive
structure.
3. FIR filters realized non recursively are always stable
4.The round off noise can be made small in non recursive realization of FIR
filters.
8. Draw the direct form realization of FIR system.
9. What is Gibbs phenomenon?
In FIR filter design by Fourier series method the infinite duration impulse response is
truncated to finite duration impulse response. The abrupt truncation of impulse response
introduces oscillators in the pass band and stop band . this effect is known as Gibbs
phenomenon.
10. Write the characteristic features of rectangular window.
1. The main lobe width is equal to 4/N
2. The maximum side lobe magnitude is -13dB
3. The side lobe magnitude does not decrease significantly with increasing
X[z] Y

[z] b
o
b
1
b
2
b
N-2
b
N-1
Z
-1
Z
-1
Z
-1
++++
11. State the condition for a digital filter to be causal and stable.
1. The digital filter transfer function H(Z) should be a rational function of Z
and the co efficient of Z should be real.
2. The poles should lies inside the unit circle in Z-plane.
3. The number of zeros should be less than or equal to number of poles.
1. When a cascade form realization is preferred in FIR filters?
The cascade from realization is preferred when complex zeros with absolute
magnitude are less than one .

2. What are the properties of FIR filters?
1. FIR filter is always stable
2. A realizable filter can always be obtained
3. FIR filter has a linear phase response.
3. For what kind of applications, the anti symmetric impulse response can be used?
The ant symmetric impulse response can be used to design Hilbert transformers and
differences.
4. Draw the frequency response of N-point rectangular window.
5. What is the response that FIR filter is always stable?
6. List the features of hamming window spectrum.
1. The mainlobe width is equal to 8/N.
2. The maximum side lobe magnitude is -41dB
3. The side lobe magnitude remains constant for increasing
7. What are the techniques of designing FIR filters?
1. Choose the desired frequency response of the filter H
d
()
2. Take inverse fourier transform of H
d
() to obtain the desired impulse
response h
d
(n).
3. Choose a window sequence W(n) and multiply h
d
(n) by W(n) to convert the
duration impulse response to finite duration impulse response h(n).
4.The transfer function H(z) of the filter is obtained by taking Z-transform of h(n).
__________________________UNIT V_________________________
1. Define multi rate DSP.
In multi rate dsp system . the samples rates are changed or are different within the
system
2. Define sub band coding
Sub-band coding (SBC) is any form of transform coding that breaks a signal into a
number of different frequency bands and encodes each one independently. This
decomposition is often the first step in data compression for audio and video signals.
.
3. What is a QMF filter?
In digital signal processing, a quadrature mirror filter is a filter most commonly
used to implement a filter bank that splits an input signal into two bands. The resulting high-
pass and low-pass signals are often decimated by a factor of 2, giving a critically sampled
two-channel representation of the original signal.
4. Define decimator.
In digital signal processing, decimation is a technique for reducing the number of
samples in a discrete-time signal.
Decimation is a two-step process:
1. Low-pass anti-aliasing filter
2. Down sampling
5. Define interpolator.
Interpolation is a method of constructing new data points within the range of a
discrete set of known data points
6. What is down sampling?
In signal processing, down sampling (or "sub sampling") is the process of reducing
the sampling rate of a signal. This is usually done to reduce the data rate or the size of the
data.

7. What is sampling rate conversion?
Sample rate conversion is the process of converting a (usually digital) signal from
one sampling rate to another, while changing the information carried by the signal as little as
possible. When applied to an image, this process is sometimes called image scaling
8. Give advantages of multi-rate DSP.
The advantages of multi rate DSP are
1.speech coding
2.image compression
3.adaptive equalization
4.echo cancellation
9. What is up sampling?
Up sampling is the process of increasing the sampling rate of a signal. For instance,
up sampling raster images such as photographs means increasing the resolution of the image.
10. Define Periodiogram.
The periodiogram is an estimate of the spectral density of a signal
11. What is the need for anti-imaging filter?
In a mixed-signal system (analog and digital), a reconstruction filter (or anti-
imaging filter) is used to construct a smooth analogue signal from the output of a digital to
analogue converter (DAC) or other sampled data output device.